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Merge branch 'audioreactive-prototype' of https://github.com/blazoncek/WLED into audio-fix

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This commit is contained in:
Blaz Kristan 2022-10-22 11:13:02 +02:00
commit ca891b0e70
3 changed files with 125 additions and 60 deletions
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175
usermods/audioreactive/audio_reactive.h
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@ -197,6 +197,7 @@ void FFTcode(void * parameter)
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
uint64_t start = esp_timer_get_time();
bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid
#endif
// get a fresh batch of samples from I2S
@ -204,9 +205,10 @@ void FFTcode(void * parameter)
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
if (start < esp_timer_get_time()) { // filter out overflows
unsigned long sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding
uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding
sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth
}
start = esp_timer_get_time(); // start measuring FFT time
#endif
xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay
@ -221,13 +223,13 @@ void FFTcode(void * parameter)
if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
}
// release highest sample to volume reactive effects early - not strictly necessary here - could also be done at the end of the function
// early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results. micDataReal = maxSample;
// early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results.
micDataReal = maxSample;
#ifdef SR_DEBUG
if (true) { // this allows measure FFT runtimes, as it disables the "only when needed" optimization
#else
if (sampleAvg > 1) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
if (sampleAvg > 0.5f) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
#endif
// run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2)
@ -255,6 +257,10 @@ void FFTcode(void * parameter)
#endif
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
haveDoneFFT = true;
#endif
} else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this.
memset(vReal, 0, sizeof(vReal));
FFT_MajorPeak = 1;
@ -267,7 +273,7 @@ void FFTcode(void * parameter)
} // for()
// mapping of FFT result bins to frequency channels
if (sampleAvg > 1) { // noise gate open
if (sampleAvg > 0.5f) { // noise gate open
#if 0
/* This FFT post processing is a DIY endeavour. What we really need is someone with sound engineering expertise to do a great job here AND most importantly, that the animations look GREAT as a result.
*
@ -325,7 +331,7 @@ void FFTcode(void * parameter)
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
if (sampleAvg > 1) { // noise gate open
if (sampleAvg > 0.5f) { // noise gate open
// Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
@ -397,16 +403,19 @@ void FFTcode(void * parameter)
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
if (start < esp_timer_get_time()) { // filter out overflows
unsigned long fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
@ -543,6 +552,7 @@ class AudioReactive : public Usermod {
// used to feed "Info" Page
unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
int receivedFormat = 0; // last received UDP sound sync format - 0=none, 1=v1 (0.13.x), 2=v2 (0.14.x)
float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
unsigned long sampleMaxTimer = 0; // last time maxSample5sec was reset
#define CYCLE_SAMPLEMAX 3500 // time window for merasuring
@ -563,21 +573,21 @@ class AudioReactive : public Usermod {
////////////////////
void logAudio()
{
if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable
#ifdef MIC_LOGGER
// Debugging functions for audio input and sound processing. Comment out the values you want to see
Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t");
//Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t");
//Serial.print("micLev:"); Serial.print(micLev); Serial.print("\t");
//Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t");
//Serial.print("sample:"); Serial.print(sample); Serial.print("\t");
Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t");
//Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t");
//Serial.print("DC_Level:"); Serial.print(micLev); Serial.print("\t");
//Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t");
//Serial.print("sampleAvg:"); Serial.print(sampleAvg); Serial.print("\t");
//Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t");
//Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t");
//Serial.print("sample:"); Serial.print(sample); Serial.print("\t");
//Serial.print("sampleMax:"); Serial.print(sampleMax); Serial.print("\t");
//Serial.print("samplePeak:"); Serial.print((samplePeak!=0) ? 128:0); Serial.print("\t");
//Serial.print("multAgc:"); Serial.print(multAgc, 4); Serial.print("\t");
Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t");
//Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t");
//Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t");
Serial.println();
#endif
@ -669,7 +679,7 @@ class AudioReactive : public Usermod {
if (time_now - last_time > 2) {
last_time = time_now;
if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
// MIC signal is "squelched" - deliver silence
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
@ -727,6 +737,7 @@ class AudioReactive : public Usermod {
else
sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value
last_soundAgc = soundAgc;
} // agcAvg()
@ -764,12 +775,13 @@ class AudioReactive : public Usermod {
micIn -= micLev; // Let's center it to 0 now
// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
float micInNoDC = fabs(micDataReal - micLev);
float micInNoDC = fabsf(micDataReal - micLev);
expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
tmpSample = expAdjF;
micIn = abs(micIn); // And get the absolute value of each sample
@ -791,6 +803,7 @@ class AudioReactive : public Usermod {
if (sampleMax < 0.5f) sampleMax = 0.0f;
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
sampleAvg = fabsf(sampleAvg); // make sure we have a positive value
} // getSample()
@ -875,11 +888,67 @@ class AudioReactive : public Usermod {
static bool isValidUdpSyncVersion(const char *header) {
return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0;
}
static bool isValidUdpSyncVersion_v1(const char *header) {
return strncmp_P(header, PSTR(UDP_SYNC_HEADER_v1), 6) == 0;
}
void decodeAudioData(int packetSize, uint8_t *fftBuff) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
// update samples for effects
volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f);
volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
// update internal samples
sampleRaw = volumeRaw;
sampleAvg = volumeSmth;
rawSampleAgc = volumeRaw;
sampleAgc = volumeSmth;
multAgc = 1.0f;
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
autoResetPeak();
if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only available on the ESP32
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f);
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
}
void decodeAudioData_v1(int packetSize, uint8_t *fftBuff) {
audioSyncPacket_v1 *receivedPacket = reinterpret_cast<audioSyncPacket_v1*>(fftBuff);
// update samples for effects
volumeSmth = fmaxf(receivedPacket->sampleAgc, 0.0f);
volumeRaw = volumeSmth; // V1 format does not have "raw" AGC sample
// update internal samples
sampleRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
sampleAvg = fmaxf(receivedPacket->sampleAvg, 0.0f);;
sampleAgc = volumeSmth;
rawSampleAgc = volumeRaw;
multAgc = 1.0f;
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
autoResetPeak();
if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only available on the ESP32
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0);
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0, 11025.0); // restrict value to range expected by effects
}
bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received.
{
if (!udpSyncConnected) return false;
bool haveFreshData = false;
size_t packetSize = fftUdp.parsePacket();
if (packetSize > 5) {
//DEBUGSR_PRINTLN("Received UDP Sync Packet");
@ -888,37 +957,17 @@ class AudioReactive : public Usermod {
// VERIFY THAT THIS IS A COMPATIBLE PACKET
if (packetSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftBuff))) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
// update samples for effects
volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f);
volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
// update internal samples
sampleRaw = volumeRaw;
sampleAvg = volumeSmth;
rawSampleAgc = volumeRaw;
sampleAgc = volumeSmth;
multAgc = 1.0f;
autoResetPeak();
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only available on the ESP32
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f);
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet");
decodeAudioData(packetSize, fftBuff);
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v2");
haveFreshData = true;
receivedFormat = 2;
} else {
if (packetSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftBuff))) {
decodeAudioData_v1(packetSize, fftBuff);
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v1");
haveFreshData = true;
receivedFormat = 1;
} else receivedFormat = 0; // unknown format
}
}
return haveFreshData;
@ -1170,9 +1219,11 @@ class AudioReactive : public Usermod {
}
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
EVERY_N_MILLIS(20) {
logAudio();
}
static unsigned long lastMicLoggerTime = 0;
if (millis()-lastMicLoggerTime > 20) {
lastMicLoggerTime = millis();
logAudio();
}
#endif
// Info Page: keep max sample from last 5 seconds
@ -1381,22 +1432,34 @@ class AudioReactive : public Usermod {
if (audioSyncEnabled) {
if (audioSyncEnabled & 0x01) {
infoArr.add(F("send mode"));
if ((udpSyncConnected) && (millis() - lastTime < 2500)) infoArr.add(F(" v2"));
} else if (audioSyncEnabled & 0x02) {
infoArr.add(F("receive mode"));
}
} else
infoArr.add("off");
if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" <i>(unconnected)</i>");
if (audioSyncEnabled && udpSyncConnected && (millis() - last_UDPTime < 2500)) {
if (receivedFormat == 1) infoArr.add(F(" v1"));
if (receivedFormat == 2) infoArr.add(F(" v2"));
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
infoArr = user.createNestedArray(F("Sampling time"));
infoArr.add(float(sampleTime)/100.0f);
infoArr.add(" ms");
infoArr = user.createNestedArray(F("FFT time"));
infoArr.add(float(fftTime-sampleTime)/100.0f);
infoArr.add(" ms");
infoArr.add(float(fftTime)/100.0f);
if ((fftTime/100) >= FFT_MIN_CYCLE) // FFT time over budget -> I2S buffer will overflow
infoArr.add("<b style=\"color:red;\">! ms</b>");
else if ((fftTime/80 + sampleTime/80) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability
infoArr.add("<b style=\"color:orange;\"> ms!</b>");
else
infoArr.add(" ms");
DEBUGSR_PRINTF("AR Sampling time: %5.2f ms\n", float(sampleTime)/100.0f);
DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime-sampleTime)/100.0f);
DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime)/100.0f);
#endif
}
}

8
wled00/wled.cpp
View File

@ -204,9 +204,11 @@ void WLED::loop()
DEBUG_PRINT(F("State time: ")); DEBUG_PRINTLN(wifiStateChangedTime);
DEBUG_PRINT(F("NTP last sync: ")); DEBUG_PRINTLN(ntpLastSyncTime);
DEBUG_PRINT(F("Client IP: ")); DEBUG_PRINTLN(Network.localIP());
DEBUG_PRINT(F("Loops/sec: ")); DEBUG_PRINTLN(loops / 30);
DEBUG_PRINT(F("UM time[ms]: ")); DEBUG_PRINT(avgUsermodMillis/loops); DEBUG_PRINT("/");DEBUG_PRINTLN(maxUsermodMillis);
DEBUG_PRINT(F("Strip time[ms]: ")); DEBUG_PRINT(avgStripMillis/loops); DEBUG_PRINT("/"); DEBUG_PRINTLN(maxStripMillis);
if (loops > 0) { // avoid division by zero
DEBUG_PRINT(F("Loops/sec: ")); DEBUG_PRINTLN(loops / 30);
DEBUG_PRINT(F("UM time[ms]: ")); DEBUG_PRINT(avgUsermodMillis/loops); DEBUG_PRINT("/");DEBUG_PRINTLN(maxUsermodMillis);
DEBUG_PRINT(F("Strip time[ms]: ")); DEBUG_PRINT(avgStripMillis/loops); DEBUG_PRINT("/"); DEBUG_PRINTLN(maxStripMillis);
}
strip.printSize();
loops = 0;
maxUsermodMillis = 0;

2
wled00/wled.h
View File

@ -8,7 +8,7 @@
*/
// version code in format yymmddb (b = daily build)
#define VERSION 2210200
#define VERSION 2210220
//uncomment this if you have a "my_config.h" file you'd like to use
//#define WLED_USE_MY_CONFIG